1. Field of the Invention
The present invention relates to an electronic musical instrument which permits easy control of harmonic coefficients in the production of a desired musical waveform by combining harmonic components while at the same time effecting level control of the respective harmonic coefficients.
2. Cross-Reference to Related Application
The present invention is related to prior art U. S. patent application Ser. No. 847,426 which matured into U.S. Pat. No. 4,700,603 on Oct. 20, 1987, entitled "Formant Filter Generator for an Electronic Musical Instrument" and assigned to the same assignee as in the subject application.
3. Description of Prior U.S. application Ser. No. 847,426 now U.S. Pat. No. 4,700,603
In the above-noted application an electronic musical instrument was proposed utilizing, in combination, a memory system and a calculation system, by which it is possible to create a format filter characteristic which enables easy control of respective harmonic coefficients through use of a simple circuit arrangement. This electronic musical instrument combines harmonic components corresponding to respective harmonic orders into the desired musical waveform having a formant filter characteristic with a harmonic order q, a cut-off harmonic order q.sub.c which is below or above the harmonic order, a level Ha, and a slope SL. The electronic musical instrument is provided with means for generating the cut-of harmonic order q.sub.c of the formant filter characteristic, means for generating the level Ha of the formant filter characteristic, memory means for storing the slope SL of the formant filter characteristic, and select means for selecting one of the level Ha from the level generating means and the slope SL from the memory means in accordance with the cut-off harmonic order q.sub.c from the cut-off harmonic order generating means. Each harmonic component value is controlled with the output signal from the selecting means.
FIGS. 3A and 3B are a block diagram illustrating the arrangement of the electronic musical instrument disclosed in the above-mentioned U.S. application. In FIGS. 3A and 3B a musical tone generating system 100 produces a desired musical tone through use of an ordinary Fourier synthesis system.
A key tablet assignor 102 scans a key tablet switch group 101 to detect the ON/OFF state, touch response, or the like of key switches included in the group 101 and holds the information of the respective switches. The information is provided to a control circuit 103 which controls the system 100.
When supplied with the information from the key tablet assignor 102, the control circuit 103 sets a composite waveform in a main memory 110 on the basis of the following Fourier sysnthesis equation (1): ##EQU1## where q is the harmonic order n the sample point number, W the number of harmonics, Cq a q-order harmonic coefficient, Fq a q-order scaling coefficient, and Zn a sample value. The procedure for the above operation is as follows: A signal is applied from the control circuit 103 to a harmonic coefficient memory 108 to read out therefrom the harmonic coefficient Cq of a timbre desired to produce. On the other hand, ADSR data which is envelope information representing temporal variations of an envelope, touch response information representing initial and after touch response data, and timbre information representing a selected timbre are applied to a scaling value generator 105, from which is obtained a scaling value Fq for scaling the harmonic coefficient Cq, i.e. a value for its level control. The harmonic coefficient Cq and the scaling value Fq are multiplied in a multiplier 107, obtaining a harmonic coefficient Cq' scaled by the scaling value Fq. The harmonic coefficient Cq' thus obtained and a q-order sine wave value, sin.sub.W.sup..pi.ng, read out of a sine wave function table 104 with a signal from the control circuit 103 are multiplied in a muliplier 106. The multiplied value from the multiplied 106 is accumulated by an accumulator 109, by which the composite waveform expressed by Eq. (1) is created and stored in a main memory 110.
Next, the composite waveform thus stored in the main memory 110 is transferred via a transfer select circuit 111 to at least one of note memories 112-1 to 112-m (where m means the provision of plural note memories, but it is evident that they can be combined into one on a time-shared basis) corresponding to keys. The composite waveform thus stored in the note memory is read out therefrom, without exerting any influence upon the synthesization of a waveform, by note frequency data from a note frequency data generator 113 which generates note frequency data corresponding to a depressed key. Data read out of the note frequency memories 112-1 to 112-m corresponding to a scale is each multiplied, in one of multipliers 114-1 to 114-m, by the envelope output waveform from an envelope generator 115 which creates an envelope waveform corresponding to each depressed key, thus producing musical waveform data added with an envelope. The musical waveform data from the multipliers 114-1 to 114-m is converted by D/A converters 116-1 to 116-m into an analog musical waveform, which is applied to a sound system 117, creating a desired musical tone.
The gist of the invention proposed in the above-mentioned application now U.S. Pat. No. 4,700,603 resides in the scaling value generator 105. The scaling value generator 105 sets the harmonic order q, the cut-off harmonic order q.sub.c, and the format filter level Ha necessary for forming the waveform, on the bases of the ADSR data, the touch response information, and the timbre information, thereby obtaining a desired formant filter characteristic.
FIG. 4 is a block diagram showing a specific operative example of the scaling value generator 105 which produces a low-pass or high-pass formant filter characteristic with resonance, and FIGS. 5A to 5D are explanatory of its operation. This is described in detail in the afore-mentioned application, and hence will be described in brief.
In FIG. 4 a subtractor 301-1 performs an operation (q-q.sub.c) and applies its output D to a complementor 302-2. The complementor 302-2 further receives, as a control input, an overflow signal Co from the subtractor 302-1 and converts the subtractor output D to an absolute value .vertline.D.vertline., which will be used as an address signal for accessing a slope memory 301. This is shown in FIG. 5A. Assuming that the chain line .circle.1 in FIG. 5B indicates the stored content of the slope memory 301, the line .circle.2 in FIG. 5B will represent the value which is read out from the slope memory 301, using the output of the complementor 302-2 as an address signal. A level comparator 308 makes a comparison between the value A of the output .circle.2 from the slope memory 301 and the value B of an arbitrarily set filter level Ha .circle.3 , and provides the comparison result to a data selector 304 via a OR gate 307. On the other hand, the output from an order comparator 303 which compares the harmonic order q and the cut-off harmonic order q.sub.c is applied to the data selection 304 via a combination of an AND-OR gate 306 and a NOT gate 305 and via the OR gate 307. The operations of these combined gates will not be described in detail for the sake of brevity. The data selector 304 selects the value A or B in accordance with the value of the harmonic order q relative to the cut-off harmonic order q.sub.c, produces a low-pass formant filter characteristic waveform shown in FIG. 5C or high-pass formant filter characteristic waveform shown in FIG. 5D, and yields the scaling value Fq corresponding to the harmonic order q. The scaling value Fq is multiplied, in the multiplier 107, by the harmonic coefficient Cq from the harmonic coefficient memory 108.
FIG. 6A shows a typical formant filter characteristic which is obtainable with the electronic musical instrument proposed in the afore-noted application. In FIG. 6A, reference character q.sub.c indicates a cut-off order which determines the cut-off position of the formant filter characteristic, Ha a level for providing the formant filter characteristic with resonsance, and SL the slope of the formant filter characteristic. The slope SL is stored in a memory. FIGS. 6B and 6C show, on an enlarged scale, the peak portion of the formant filter characteristic depicted in FIG. 6A. In FIG. 6B, the full line shows the current cut-off order q.sub.c on an enlarged scale. When the cut-off order changes from q.sub.c and q.sub.c ', even if an address for reading out the slope memory is generated on the basis of the cut-off order expressed by an integer alone or an integer and a decimal, the cut-off ouder will undergo an abrupt change from q.sub.c to q.sub.c ' as seen from a change from the full line to the broken line in FIG. 6B, in case the slope memory does not store data including the decimal part of the cut-off order.
Such an abrupt change of the cut-off order leads to the generation of a very jarring, temporarily-varying noise in the musical tone that will ultimately be produced. This problem could be solved by improving the resolution for slope data so that the cut-off harmonic order varies, little by little, from q.sub.c (indicated by the solid line), to q.sub.c ' as indicated by the broken line in FIG. 6C. With such finer resolution, the temporarily-varying noise could be reduced. One possible method for obtaining greater resolution is to increase the capacity of the slope memory, but this is uneconomical. In order to raise the resolution by a precision of 3 to 4 bits, for example, the memory capacity myst be increased 8 to 16 times. The present inventor has succeeded in implementing this function by interpolating values stored in the slope memory, without increasing its capacity.